Digital Subscriber Line (DSL, Digital Subscriber Loop, xDSL) involves a technology that enables high-speed transmission of digital data over traditional copper telephone lines. This technology involves digital telecommunications protocols designed to allow high-speed data communication over existing copper telephone lines between end-users and telephone companies.
When two conventional modems are connected through the telephone system (e.g., Public Switched Telephone Network (PSTN)), the communication may be treated the same as voice conversations. This has the advantage that there is no investment required from the telephone company (telco) but the disadvantage is that the bandwidth available for the communication is the same as that available for voice conversations, usually 64 kb/s (DSO) at most. The twisted-pair copper wires into individual homes or offices can usually carry significantly more than 64 kb/s, provided the telco handles the signal as digital rather than analog.
There are many implementations of the basic scheme, differing in the communication protocol used and providing varying service levels. The throughput of the communication can be anything from about 128 kb/s to over 8 Mb/s, the communication can be either symmetric or asymmetric (i.e., the available bandwidth may or may not be the same upstream and downstream). Equipment prices and service fees also vary considerably.
In many different kinds of modem telecommunications equipment, an important element is a voice processing subsystem, which may perform such functions as transcoding, Dual Tone Modulation Frequency (DTMF) processing, echo cancellation, etc. Examples of equipment requiring voice processing of this kind include everything from speakerphones, to Global System for Mobile communications (GSM) basestations, to broadband integrated access devices. Voice processing subsystems may be Digital Signal Processing (DSP) based and feature a set of algorithm implementations in software. These algorithms may be hand-coded in assembly-code form by algorithmic and DSP-programming experts. Also, an easy way to combine the required algorithms in the required combinations and then interface to the voice processing subsystem through a simple external interface is desired.
Voice over Digital Subscriber Line (VoDSL) involves leveraging copper infrastructure to provide quality voice services and support a wide variety of data applications over an existing line to a customer. VoDSL implements DSL platform in conjunction with platform adaptations that enable voice services. It further gives data competitive local exchange carriers (CLECs) a way to increase revenue potential, incumbent local exchange carriers (ILECs) an answer to the cable modem, and interexchange carriers (IXCs) a way to gain access to the local voice loop. Thus, any carrier type may increase the value of services available through VoDSL.
Generally, VoDSL involves a voice gateway, an integrated access device (IAD), among other components. The voice gateway may provide voice packets that are depacketized and converted to a format for delivery to a voice switch or other similar device. The voice gateway may enable traffic to be accessed from a data network and forwarded to PSTN for service and switching. The IAD may serve as a DSL modem and perform other functionality. The IAD may serve as an interface between a DSL network service and a customer's voice and data equipment. The IAD may provide the interface between the DSL network service and a customer's network equipment. Further, an IAD may be used to connect voice and data enabled equipment.
VoDSL may also be transmitted via Internet Protocol (IP). VoIP may be defined as voice over Internet Protocol, which includes any technology that enables voice telephony over IP networks. Some of the challenges involved with VoIP may include delivering the voice, fax or video packets in a dependable manner to a user. This may be accomplished by taking the voice or data from a source where it is digitized, compressed due to the limited bandwidth of the Internet, and sent across the network. The process may then be reversed to enable communication by voice. VoIP enables users, including companies and other entities, to place telephony calls over IP networks, instead of PSTN.
A consideration associated with the use of VoDSL, VoIP and other voice applications involves silence suppression which may be used to enhance bandwidth and throughput. Silence suppression removes the necessity of packetizing the silence portion of a phone conversation (e.g., when no one is talking). To optimize bit-rates in simultaneously transmitting voice and data information, a voice signal detector detects silence portions of the speech signal. Rather than transmit the silence portion of the voice signal, data (e.g., silence insertion descriptor) may be inserted into the packet stream thereby recovering bandwidth that would otherwise be allocated for voice traffic. While providing effective bit-rate reduction, the deletion of background noise that typically accompanies the “silence” portions of the voice data has the undesired effect on the person receiving and listening to the voice data of absolute silence and the perception of on/off transmission rather than a continuous connection.
In conjunction with silence suppression, comfort noise generation may be implemented to reconstruct or construct and replace the silence part of speech and other voice signals. A drawback associated with conventional comfort noise generators is that they require a large MIPS (million instructions per second) and memory capacity and reduce efficiency and effective voice transmission.
Existing International Telecommunications Union (ITU) recommendation G. series G729AB uses a simpler approach for the gaussian noise generation, which has the drawback of periodicity. Other generators are more MIPS intensive and are not generally suitable for real time systems or the complexity is not warranted.
Gaussian white noise generators may be implemented in applications involving synthesizing speech and other voice signals. One of the ways in which the gaussian generator may be implemented may include using a central limit theorem on a uniform random generator. However, this has a drawback of periodicity especially when dealing with the long-term generation of constant amplitude speech, noise signal or other applications. Other generators are more MIPS intensive and are not generally suitable for real time systems or the complexity is not warranted.
Typically there are very tight latency requirements on telecommunications devices, as excessive latency degrades the quality of a telephone conversation. Consequently, signal processing algorithms used in telecommunications often have to execute on very small blocks of voice data. For example, in VoDSL Customer Premise Equipment (CPE), the Digital Signal Processor operates on 4 sample blocks of 8 kHz data.
An advanced feature of voice compression in voice over data network systems is adaptive silence compression and reconstruction. One aspect of this feature is that a simulated background noise signal is generated by filtering white gaussian noise with a filter intended to spectrally shape the noise to closely match a ‘true’ background noise, which was not transmitted in order to save bandwidth.
The filter coefficients, however, do not necessarily contain the correct gain, so the resultant signal is not the same power as the true background noise. Also the excitation to the filter generally has some gain which causes the output to be of a different gain from that of the true background noise. In addition, an efficient generation of the simulated signal may only generate four samples at a time, making it difficult (and computationally expensive, given that this function is called approximately 2000 times per second) to measure the signal strength and compensate the gain accordingly.
Therefore, there is a need in the art of VoDSL and VoIP for a more efficient method and system for transmitting voice signals.